Method for management of voice-over IP communications

ABSTRACT

Method for admitting new voice calls to a VoIP network. Various points in the network are polled to generate a link utilization parameter. Parameters denoting threshold values of link utilization in the network are compared to the link utilization parameter. A first comparison is made between a first (current) link utilization parameter and first and second link utilization threshold parameters. Then, a second comparison is made between the current link utilization parameter and a second (previous) value of link utilization. A calculation of new calls allowed into the network is performed based on these comparisons and whether the current level of link utilization is rising or falling when compared to a previous level of link utilization.

FIELD OF THE INVENTION

The invention relates to the field of communications systems and morespecifically to the Quality of Service (QoS) management and admissioncontrol for voice calls of varied classifications in a voice-over IP(VoIP) network.

DESCRIPTION OF THE BACKGROUND ART

Traditionally, voice calls are transported entirely over the end-to-end,circuit-based Public Switched Telephone Network (PSTN). However,considerable attention has been directed toward the implementation ofreal-time communication across computer data networks, and particularlythe ability to route voice traffic over these networks. Interest hasalso been raised in using Voice over IP (VoIP) solutions to facilitatevoice communication between originating and terminating PSTN end pointsand enterprise or private network end points served by PSTN Switches,Private Branch Exchanges (PBX), or IP end points in Local Area Networks(LAN) via the Internet or private IP network. Using a private IP networkor Internet for long haul routing substantially bypasses the PSTN. Whenbypassing a PSTN, pulse code modulated (PCM) voice traffic is processedinto IP (or ATM) packets, transported over the private IP network orInternet (or ATM network), and then processed back to PCM voice. Tofacilitate such call routing, originating and terminating End Office(EO) switches are connected to PSTN/IP (or PSTN/ATM) gateways thatreside as hosts on the IP (or ATM) network.

Unfortunately, when a new VoIP telephone voice call is established (withthe intent of it being routed over the IP network), there are no meansto evaluate the level of congestion of the core IP network. In otherwords, it is possible to have too many new voice calls being introducedto the network at the same time so that the core IP network isoverloaded. Under such a condition, it is highly likely that packets ofinformation that contain the voice data will either be dropped, ordelayed from arriving at the destination. These conditions result inpoor Quality of Service (QoS) of the network.

The problem of QoS is further compounded when the communication networkmust handle voice calls of different priority levels or classifications,which exist, for example, in a Defense Switched Network (DSN). That is,there is a need to provide a differentiation in quality based on thepriority level of the call as well as provide quality to all calls inthe system. The highest priority calls, for instance, must not beblocked, and must receive the best voice quality, even during trafficoverloads and IP network congestion. These types of conditions can ariseduring crisis or partial network failures. For example, existingarchitectures tend to drop established calls (i.e., calls of lowerprecedence) to increase network resources for use by higher prioritycalls. This process is disruptive and not user-friendly.

SUMMARY OF THE INVENTION

Various problems in the prior art are addressed by a method fordetermining how to admit to a network, employing VoIP, new calls basedon link utilization of various links in the network. Parameters denotingthreshold values of link utilization in the network are compared toparameters indicative of actual link utilization levels. A calculationof what fraction of new calls can be allowed into the network isperformed based on these comparisons and whether the current level oflink utilization is rising or falling when compared to a previous levelof link utilization. This two-fold aspect of making comparisonsincreases the flexibility of the network to adapt to changing conditionstherein.

More specifically, in one embodiment of the invention, various links inthe network are polled to determine their corresponding level of linkutilization to generate a link utilization parameter. A first comparisonis made between a first (current) link utilization parameter and firstand second link utilization threshold parameters. In one example, thefirst and second link utilization threshold parameters denote a low linkutilization threshold and a high link utilization threshold,respectively. Depending upon the results of the comparisons, either anallowed call value is assigned to the network or a second comparison ismade. The second comparison is made between the current link utilizationparameter and a second (previous) value of link utilization. Dependingupon the results of this second comparison, the allowed call value iseither adjusted to compensate for an increase or decrease in linkutilization or the allowed call value remains unchanged from it previousvalue. The allowed call value can either be implemented as a fraction ofall call attempts or as a fixed rate. New calls are subsequentlyadmitted to the network based upon the allowed call value rather thanhave a continuing number of calls enter the network at the expense of a“dropped” call elsewhere in the network to provide the necessarybandwidth. The method can be practiced in a variety of differentnetworks including PSTN-based systems with IP-related componentsinterfaced therein, end-to end IP networks and networks having to serveusers with different levels of network usage priority or authorizationlevels.

BRIEF DESCRIPTION OF THE DRAWINGS

The teachings of the present invention can be readily understood byconsidering the following detailed description in conjunction with theaccompanying drawings, in which:

FIG. 1 depicts a first system architecture for practicing VoIP calladmission in accordance with an embodiment of the subject invention;

FIG. 2 depicts a second system architecture for practicing VoIP calladmission in accordance with an embodiment of the subject invention;

FIG. 3 depicts a series of method steps for practicing VoIP calladmission in accordance with an embodiment of the subject invention;

FIG. 4 depicts detailed view of call admission and call managementmodules associated with an embodiment of the subject invention;

FIG. 5 depicts a chart of link utilization and dynamic policy states inaccordance with an embodiment of the subject invention;

FIG. 6 depicts a graph of link utilization measurement vs. blockingprobability for a first network condition when executing an embodimentof the subject invention, and

FIG. 7 depicts a graph of link utilization measurement vs. blockingprobability for a second network condition when executing an embodimentof the subject invention.

To facilitate understanding, identical reference numerals have beenused, where possible, to designate identical elements that are common tothe figures.

DETAILED DESCRIPTION

The subject invention is adapted to assist in establishing and/ormanaging of VoIP traffic in a network (for example an Internet Protocol(IP) network) by, illustratively, monitoring criteria indicative of theimportance of a new voice call entering the network, and networkcapacity and the like. Accordingly, exemplary telecommunications systemsare described as potential environments in which the subject inventionmay be utilized.

FIG. 1 depicts an exemplary telecommunications system 100 for routingtelephone calls between at least a first wire line subscriber 102 and atleast a second wire line subscriber 104. The telecommunication system isan enhanced PSTN-based system. That is, telephone calls are routedacross a supplementary intermediate data network 118 implementing anetwork layer protocol and/or (such as IP) a link layer protocol (suchas asynchronous transfer mode (ATM) instead of or in addition to aseries of previously existing switches).

The telecommunications system 100 of FIG. 1 includes a first subscriberend office unit 106 connected to the first subscriber 102 and a secondend office 108 connected to the second subscriber 104 via conventionallocal loop subscriber lines (103 and 105, respectively). For example,such first subscriber line 103 and second subscriber line 105 wouldtypically be implemented using two-element twisted pair wires carryinganalog information or basic rate ISDN digital information.

Communication paths are established using the data network 118 andbypassing some PSTN related components by connecting the first endoffice 106 to a first gateway 114 and likewise connecting the second endoffice unit 108 to a second gateway 116. First and second gateways 114and 116 respectively reside as hosts on the network 118. The gatewaysenable VoIP services to the first wire line subscriber 102 and secondwire line subscriber 104 communicating over the network 118. Whenutilizing the data network 118 between the first wire line subscriber102 and the second wire line subscriber 104, traffic is routed from thefirst end office 106 and second end office 108 to the respectivegateways 114 and 116 for routing across various locations or nodes 120in the data network 118.

A first softswitch 110 is connected to the first subscriber end officeunit 106 and first gateway 114. A second softswitch 112 is connected tothe second subscriber end office unit 108 and second gateway 116. Thesoftswitches 110 and 112 coordinate with their respectively connectedgateways to synchronize the signals requested to start data transferfrom the gateways to the data network 118. A Poller 124 is connected tothe various nodes 120 in the network 118 and to the softswitches 110 and112. Within the Poller 124 is a Call Admission Manager (CAM) 126. Withineach softswitch 110 and 112 is a Call Admission Controller (CAC) 128.Depending on the implementation, the CAC 128 can be a standalone deviceor may be implemented within the gateways 114 and 116. The CAC's 128 andCAM 126 each execute specific and dedicated algorithms in order tomonitor the status of links in the network 118 and admit new calls tothe network accordingly. The details of CAC and CAM algorithms aredescribed in David Houck and Gopal Meempat, “Call admission control andload balancing for voice over IP”, Performance Evaluation Vol. 47, No.4, March 2002, Pages: 243-253 herein incorporated by reference in itsentirety.

FIG. 4 details the internal circuitry of the softswitches 110 and 112,as well as the poller 124. Specifically, the softswitches 110 and 112and poller 124 each comprise at least one central processing unit (CPU)130, support circuits 134, and memory 136. The CPU 130 may comprise oneor more conventionally available microprocessors. The support circuits134 are well known circuits that comprise power supplies, clocks,input/output interface circuitry and the like. Memory 136 may compriserandom access memory, read only memory, removable disk memory, flashmemory, and various combinations of these types of memory. The memory136 is sometimes referred to as main memory and may in part be used ascache memory or buffer memory. The memory 136 stores various softwarepackages 132 and 138 that dictate call admission policies in the CAC 128and CAM 126 modules respectively.

FIG. 2 depicts an exemplary end-to-end VoIP network system 200 (i.e.,there are little or no PSTN-related components). The end-to-end VoIPnetwork system 200 has similarities to the PSTN based telecommunicationssystem 100 and the specific distinctions are described below.Specifically, instead of first and second wire line subscribers 102 and104, the end-to-end VoIP system 200 contains first IP customer 202 andsecond IP customer 204. Each of said IP customers 202 and 204,respectively, will have one or more customer devices, which may beselected from the group consisting of an IP phone, an IP soft clientbased component in a laptop, or other wired or wireless communicationdevice 202 _(n) and 204 _(n). Each of the IP customer devices are linkedto the intermediate data network 118 via a local first customer networkrouter 206 and a second customer network router 208, respectively. Eachof the customer network routers 206 and 208, respectively, are connecteddirectly to the intermediate data network 118 via first edge router 210and second edge router 212, respectively. The first edge router 210 andsecond edge router 212 are logically connected to the Poller 124. Insimilar fashion to the PSTN-based communications network 100,softswitches 110 and 112 are logically connected to the Poller 124 aswell as to subscriber components such as IP subscriber components 202and 204.

In either of the communication systems 100 or 200, a new voice call isadmitted into the network via a path and at a time when the congestionlevels that exist in the network are sufficiently below system limits toallow the call and to allow it at a sufficient data rate to providecontinuing quality of service for the new call as well as all existingcalls in the network. In one embodiment, the CAC 128 and CAM 126 areresponsible for managing these tasks as described in the above-citedreference and also as discussed in co-pending patent application Ser.No. 10/674,885 filed Sep. 30, 2003 and Ser. No. 10/674,123 filed Sep.26, 2003 herein incorporated by reference in its entirety.

The basic theory of operation of the subject invention is depicted inthe chart 500 of FIG. 5. In general, a measurement of link utilizationin the network (i.e., for every link) is determined. Based on the linkutilization and whether such utilization is increasing or decreasing, adetermination is made as to the rate at which new voice calls areadmitted into the network. The rate is expressed, in one embodiment, asa fraction of all new calls attempting to enter the network. The chart500 is a measure of the current link utilization in the network and isdivided by two boundaries to define three decision regions.Specifically, a first region 502 and a second region 504 are defined as,respectively, regions below and above a first threshold level 508.Similarly, second region 504 and a third region 506 are defined as,respectively, regions below and above a second threshold value 510.

In one embodiment of the invention, first threshold value 508 is definedas a low threshold, wherein if current link utilization is below the lowthreshold all new calls coming into the network will be admitted,thereby defining a policy of the first region 502. In such anembodiment, the second threshold 510 is defined as a high threshold,wherein if current link utilization is above the high threshold value, adecrease in the allowed rate of new calls entering the network occurs ifthe link utilization is rising, thereby defining third decision region506. If current link utilization is between the lower threshold 508 andthe high threshold 510, an increase in the allowed new call rate intothe network is permitted if the link utilization is falling, therebydefining second decision region 504. The conditional aspects of secondregion 504 and third region 506 allow for greater flexibility in thecall admission policy of the network. That is, call admission policiesare not dictated strictly upon a set value or threshold level of linkutilization, but also upon whether the link utilization rate is risingor falling. This provides for an extra level of flexibility andadaptability within the network so that an optimum number of new callscan be administered by the network.

As discussed earlier with respect to FIG. 5, the subject inventionoperates in one embodiment to increase the blocking probability of a newcall when the congestion is severe, as defined by the current linkutilization being higher than a “high threshold.” When the linkutilization is less than a “low threshold,” all call set-up requests areaccepted. When the link utilization is between the two thresholds, theblocking probability is reduced if the link utilization is falling.Depending on the movement of the utilization level, the blockingprobability can also be kept constant. This methodology applies to allaccess network technologies including DSL, cable, Ethernet, and wirelessas long as there is access control via Softswitch or some other vehicle.

In greater detail and with specific regard to the subject invention,FIG. 3 depicts a series of method steps 300 that the CAM 126 follows toestablish a dynamic call policy for the softswitches 110 and 112 tohandle calls. The method 300 requires evaluation of a plurality ofparameters in accordance with, illustratively, the chart 500 of FIG. 5.Table 1 identifies a plurality of dynamic CAM policy parameters andtheir definitions suitable for use in the present invention. Otherpolicy parameters may also be employed. TABLE 1 Allowed_Frac Allowedfraction of the offered load = 1 − Blocking Probability Current_UtilCurrent utilization measurement Previous_Util Previously measuredutilization Low_Threshold Lower bound of congested link utilizationHigh_Threshold Upper utilization bound to apply additive increase MultFixed multiplicative decrease factor Addi Additive increase factor

The method 300 starts at step 302 and proceeds to step 304, where theCAM 126 receives polled information from various locations 120 in thenetwork 118 regarding link utilization for voice for all links. In oneembodiment, polling is performed at intervals in the range ofapproximately 10-60 seconds and preferably approximately every 30seconds.

Once this information is received, step 304 executes a decision whereinthe current link utilization value is compared to the low thresholdvalue. If the current utilization value is less than the low thresholdvalue, the method proceeds to step 306 where the dynamic CAM policyparameter Allowed_Frac is set equal to 1 (which effectively means thatno calls are blocked from entering the network). If the currentutilization value is greater than the low threshold value, the methodproceeds to step 308.

At step 308, the current link utilization value is compared to the highthreshold value. If the current link utilization value is higher thanthe high threshold value, then the method proceeds to step 310. At step310, a determination is made as to whether the current link utilizationvalue is greater than the previous link utilization value or if thecurrent link utilization value is greater than or equal to 100% of thelink utilization capacity of the network. Utilization values greaterthan 100% are possible since this measure includes dropped packets. Ifthe current link utilization value does not satisfy either of theconditions of decision step 310, the method proceeds to step 316 wherethe dynamic CAM parameter Allowed_Frac retains its current value. If thecurrent link utilization value satisfies the requirements of decisionstep 310, the method proceeds to step 312 where the dynamic Camparameter value Allowed_Frac is evaluated as follows:Allowed_(—) Frac=Allowed_(—) Frac*(Mult−(Current_(—) Util−High_(—)Thres))

Returning to step 308, if the current link utilization value is notgreater than the high threshold level value, then the method proceeds tostep 314 where, the current link utilization value is compared to theprevious link utilization value. If the current link utilization valueis not less than the previous link utilization value, the methodproceeds to step 316 where, as indicated earlier, the value of thedynamic CAM parameter Allowed_Frac remains as it currently is. If thecurrent link utilization value is less than the previous linkutilization value, the method proceeds to step 318 where the value ofAllowed_Frac is updated as follows:Allowed_(—) Frac=Allowed_(—) Frac+AddiThe method ends as step 320.

The CAM 126, in effect, gathers information from the network routersproviding link and process utilizations that enables a call blockingpolicy (the throttling mechanism for the incoming calls based on thelink utilization) in the CAC. Since the CAM 126 is a network function,it also enables changing the blocking policy variables via networkmanagement interfaces. The function of the CAC 126 is distributed in thesoftswitches 110/112 or gateways 114, 116 where it is either integratedinto the call processing logic or hosted on an adjunct processor or isstandalone implemented on a server. More specifically, the role of theCAM 126 is to periodically poll IP routers in the network 118 using, forexample, SNMP to receive utilizations of the voice traffic class at eachlink. The CAM 126 is aware of existing MPLS paths throughout the network118. This information can also be collected through SNMP polling. Thus,the CAM 126 can relate a congested link to a set of network paths thatuse the congested link. Hence, whenever a congested link is detected,the CAM 126 prepares a policy for all paths going through this link, andsends the policy to the CAC 128, which is located in a one of the softswitches 110/112. For example, a policy can be “Block x % of new callset-up requests between network ingress A and network egress B.” A CACdatabase in a particular softswitch contains blocking policy only for asubset of paths, the ones involving gateways/edge routers controlled bythat softswitch. During call processing, the softswitch 110/112 looksinto the CAC database to determine if the call is to be allowed orblocked. Alternately, the CAM 126 can send link status information(acquired prior to step 304) and let the CAC 128 perform the policydecision that is best for the current update period.

There is no per-call interaction between the softswitch 112 and the CAM126, or per-call computations. The logic of the CAC/CAM algorithmsfollows these three steps, executed in sequence, once every T seconds(referred to as the update interval):

-   1. Update the link utilization database in the CAM 124. This is    accomplished by receiving traffic measurement reports for all IP    routers, once every T seconds.-   2. Compute the admission control and load balancing decisions to be    used by the softswitches 112 until the next update.-   3. Disseminate the control decisions to the softswitches 112 where    they are executed for each new call arrival, until the next update    is received.

It should be noted that these controls can be computed and disseminatedasynchronously. That is, the CAM can poll a link's status, and ifchanged, can immediately update the path's policy that use that link andsend out a new policy to the softswitch. There is no need to wait topoll all the links first.

As an additional feature, each new call entering the network canoptionally be assigned a call-level and a packet-level priority basedupon a priority scale. As this new call enters the communication system100 or 200, said call is processed in such a manner as to evaluate itspriority level and its destination to determine whether said call haspriority over an existing call already at the destination. Based on saidnew call priority, the quality of said new call is also processed sothat a higher priority call receives better quality of service than acall of lower priority already existing on the network. New originatingcalls of higher priority reaching a destination with an existingin-progress call of lower relative priority will also preempt said lowerpriority existing call. Details of the call prioritization are presentedin U.S. patent application Ser. No. 10/674,123 filed Sep. 26, 2003 asidentified above.

In detail, management of new voice calls in accordance with the subjectinvention utilizes both packet- and call-level controls. At the packetlevel, Differentiated Services (DiffServ) and Multi-Protocol LabelSwitching (MPLS) technologies assign multiple priorities or levels tovoice calls and isolate voice traffic from data and other traffic types.DiffServ is generally discussed in “An Architecture for DifferentiatedServices” by Blake, et al., RFC2475, December, 1998 and MPLS isgenerally discussed in “Multiprotocol Label Switching Architecture” byRosen et al, RFC 3031, January 2001, both herein incorporated byreference in their entireties. For example, calls above a “Routine”level do not experience any call blocking. They receive prioritytreatment from the routers within the network; thus, experience little,if any, packet loss. The “Routine” calls experience call blocking duringnetwork congestion. The Diffserv marking of the packets reflects packetlevel priorities; hence, the routers can offer preferential treatmentfor high priority call types. This guarantees that even during heavytraffic overloads, the packets of priority calls receive adequateservice. The “Routine” calls may experience small amounts of packetloss; however, VoIP calls can tolerate packet loss values of 1-3%. Foremergency purposes, even 5-10% packet loss is considered acceptable.

This elasticity in packet network capacity leads to a new class ofadmission control algorithms. Different priorities in the packet-levelare implemented by putting voice traffic into a Diffserv AssuredForwarding (AF) service class and using a drop precedence to distinguishpriority levels. As an example, by using the AF class formulti-precedence voice in our solution, one can use the AF1 serviceclass for voice with all voice priority classes sharing the same queueand bandwidth. From there, different drop precedence markings could beused to distinguish the priority classes as shown in Table 2. Eachpriority level could have different lower and upper thresholds S08 andS10 of FIG. 5 to further optimize the network. TABLE 2 Priority LevelMarking Flash and Flash Override AF11 Intermediate and Priority AF12Routine AF AF13Since there are relatively few high priority calls entering a givennetwork, the high priority calls will rarely lose packets and receiveexcellent voice quality.

Call-level control mechanisms are activated to limit additional voicetraffic into congested links. This is achieved through the novelmeasurement-based admission control algorithm that combines usefulfeatures of Diffserv and MPLS. The framework is based on using DiffServto allocate bandwidth for voice traffic on each link and MPLS to specifythe routing of voice packets. This way, when congestion is detected at agiven link, one can block new set up requests for calls whose packetswould go through the congested link.

The policies received from the CAM 126 are applied by the CAC 128 untilnew policies are received. Note that in one embodiment of the invention,these policies are only applied to the “Routine” level calls. However,different policies can be applied to other call levels, i.e.,Intermediate and Priority, for added system flexibility. This frameworkenables high precedence calls to “soft-preempt” lower precedence callson congested links. For example, when a high precedence call is admittedinto a congested link, the “Routine” precedence traffic experienceshigher delays and possibly higher packet losses while the highprecedence traffic receives sufficient service since this traffic iscarried using higher priority packets. Since new “Routine” calls will beblocked under severe congestion, any overload situation will bealleviated quickly. This is superior to dropping an already existing lowprecedence call immediately as is done in the PSTN networks, since thelow precedence users can tolerate some performance degradation.

The subject invention is designed for flexibility in that it will workin different types of communication networks such as those describedabove. If the network in which the subject invention operates isdedicated exclusively for voice traffic, the method proceeds in themanner described above. If the communication network has different typesof traffic (i.e., voice traffic and data traffic) traveling along thesame communication paths, the invention operates along the serviceclasses reserved for voice traffic to practice the invention.Specifically, in multi-traffic environments, a classification system(for example, DiffServ) is used to separate the voice traffic from thedata traffic. The two types of classes used are identified in theStandards IETF RFC 2998 “A Framework for Integrated Services Operationover Diffserv Networks”, Y. Bernet, P. Ford, R. Yavatkar, F. Baker, L.Zhang, M. Speer, R. Braden, B. Davie, J. Wroclawski, E. Felstaine.November 2000; IETF RFC 3246 “An Expedited Forwarding PHB (Per-HopBehavior)”, B. Davie, A. Charny, J. C. R. Bennet, K. Benson, J. Y. LeBoudec, W. Courtney, S. Davari, V. Firoiu, D. Stiliadis. March 2002; RFC3260 “New Terminology and Clarifications for Diffserv”, D. Grossman.April 2002; IETF RFC 2597 and IETF RFC 2598 as Assured Forwarding (AF)and Expedited Forwarding (EF) PHB. Those skilled in the art will readilynote that the classifications indicated are but one example of howdifferent types of traffic can be assigned to DiffServ classes.Depending on the applications and their requirements in themulti-traffic network, different assignments will best serve insatisfying the QoS requirements.

Results of the dynamic policy algorithm are shown in FIGS. 6 and 7.Specifically, FIG. 6 depicts a graph 600 of link utilization andblocking probabilities for a specific number of measurements (pollingintervals) when the network utilizing the dynamic policy is at theengineered load. Similarly, FIG. 7 depicts a graph 700 also depictingutilization and blocking probabilities versus measurements when thenetwork utilizing the dynamic policy is at 100% overload. In each of thegraphs 600 and 700 the upper most curve 602/702 respectively measureslink utilization, while the lower curve 604/704 respectively measuresblocking probability. When the policy is in operation in a network thatis operating at its expected or engineered load per FIG. 6, as thenumber of new calls attempting to enter the network increases thusincreasing link utilization, there is a spike at point 606A in theengineered load curve which, via the dynamic policy algorithm, resultsin a corresponding increase in blocking probability of calls at point606B. Conversely as the blocking probability takes effect and less callsare introduced to the network, there is a decrease in the linkutilization, such as at point 608A, which results in a correspondingdrop in the necessary blocking probability at point 608B. Similar causeand effect relationships are seen in FIG. 7 when the network is at 100%overload. For example, as link utilization temporarily exceeds a certainvalue at point 706A, there is a corresponding increase in the blockingprobability as seen at point 706B. Conversely, as the link utilizationdrops below a certain level at point 708A, there is a correspondingdecrease in the blocking probability of new calls entering as seen atpoint 708B. The primary difference between curves 604 and 704 being thatcurve 704 is at a relatively higher blocking probability because thenetwork managing the new voice calls is already at an overloadcondition; thus, operating at a higher blocking probability level ingeneral. For each of the graphs, the measurements are in the form ofpolling intervals and in these particular examples, show the results of200 polling intervals which is approximately 100 minutes of networkoperation.

Although various embodiments which incorporate the teachings of thepresent invention have been shown and described in detail herein, thoseskilled in the art can readily devise many other varied embodiments thatstill incorporate these teachings.

1. A method of voice over IP call admission in a network, comprising:(a) receiving a first parameter indicative of a level of linkutilization in said network; (b) comparing said first parameter to atleast one threshold parameter indicative of link utilization capacity;(c) comparing said first parameter to a second parameter indicative of aprevious level of link utilization in the network when the firstparameter exceeds a first of the at least one threshold parameters; and(d) determining an allowable call value in response to said parametercomparisons.
 2. The method of claim 1 wherein the information isreceived from polling one or more locations in the network.
 3. Themethod of claim 1 wherein the network is an enhanced PSTN network. 4.The method of claim 3 wherein the enhanced PSTN network furthercomprises at least two IP gateways interfacing with respective PSTNendpoints and an intermediate IP network.
 5. The method of claim 1 wherethe network is an end-to-end IP network.
 6. The method of claim 1wherein the step of comparing said first parameter to at least onethreshold parameter indicative of link utilization capacity furthercomprises comparing the first parameter to the first threshold parameterindicative of link utilization capacity and a second threshold parameterindicative of link utilization capacity.
 7. The method of the claim 6where the first threshold parameter indicative of link utilizationcapacity is a low threshold and the second threshold parameterindicative of link utilization capacity is a high threshold.
 8. Themethod of the claim 7 wherein the low threshold parameter has a value ofapproximately 88% of the capacity of the link utilization.
 9. The methodof the claim 7 wherein the high threshold parameter has a value ofapproximately 96% of the capacity of the link utilization.
 10. Themethod of claim 1 wherein the step of comparing said first parameter toa second parameter indicative of a previous level of link utilization inthe network if the first parameter exceeds a first of the at least onethreshold parameters further comprises determining if the firstparameter is less than a value of a parameter indicative of a previouslevel of link utilization.
 11. The method of claim 10 wherein if thefirst parameter is less than a value of a parameter indicative of aprevious level of link utilization, then the allowable call value iscalculated as:Allowed_(—) Frac=Allowed_(—) Frac+Addi where Addi is an additive factorto increase the allowable fraction of calls.
 12. The method of claim 10wherein if the first parameter is not less than a value of a parameterindicative of a previous level of link utilization, then the allowablecall value remains at its current value.
 13. The method of claim 10wherein the step of comparing said first parameter to a second parameterindicative of a previous level of link utilization in the network if thefirst parameter exceeds a first of the at least one threshold parametersfurther comprises determining if the first parameter is greater than avalue of a parameter indicative of a previous level of link utilizationor greater than or equal to 100% of link utilization.
 14. The method ofclaim 13 wherein if the first parameter is greater than a value of aparameter indicative of a previous level of link utilization or greaterthan or equal to 100% of link utilization, then the allowable call valueis calculated as:Allowed_(—) Frac=Allowed_(—) Frac*(Mult−(Current_(—) Util−High_(—)Thres)) where Mult is a multiplicative decrease factor, Current_Util isthe first parameter and High_Thres is one of the threshold parameters.15. The method of claim 13 above wherein if the first parameter is notgreater than a value of a parameter indicative of a previous level oflink utilization or not greater than or equal to 100% of linkutilization, then the allowable call value remains at its current value.16. A computer readable medium containing a program which, whenexecuted, performs an operation of admitting of voice over IP calls in anetwork the operation comprising: (a) receiving a first parameterindicative of a level of link utilization in said network; (b) comparingsaid first parameter to at least one threshold parameter indicative oflink utilization capacity; (c) comparing said first parameter to asecond parameter indicative of a previous level of link utilization inthe network when the first parameter exceeds a first of the at least onethreshold parameters; and (d) determining an allowable call value inresponse to said parameter comparisons.
 17. The computer readable mediumof claim 16 wherein the step of comparing said first parameter to asecond parameter indicative of a previous level of link utilization inthe network if the first parameter exceeds a first of the at least onethreshold parameters further comprises determining if the firstparameter is less than a value of a parameter indicative of a previouslevel of link utilization.
 18. An apparatus for facilitating voice overIP call admission in a network, comprising: means for receiving a firstparameter indicative of a level of link utilization in said network;means for comparing said first parameter to at least one thresholdparameter indicative of link utilization capacity; means for comparingsaid first parameter to a second parameter indicative of a previouslevel of link utilization in the network when the first parameterexceeds a first of the at least one threshold parameters; and means fordetermining an allowable call value in response to said parametercomparisons.
 19. The apparatus of claim 18 wherein the means forreceiving further comprises a Call Admission Manager Module (CAM) incommunication with at least one link in the network. 20 The apparatus ofclaim 19 wherein the CAM receives information by polling said at leastone link in the network.
 21. An apparatus for facilitating voice over IPcall admission in a network, comprising: means for receiving calladmission policy information based upon a first parameter indicative ofa level of link utilization in said network relative to at least onethreshold parameter indicative of link utilization capacity and a secondparameter indicative of a previous level of link utilization in thenetwork when the first parameter exceeds a first of the at least onethreshold parameters; and means for generating an allowable call valueas a function of said first parameter being compared to said at leastone threshold parameter and said second parameter to arrive at the calladmission policy information.
 22. The apparatus of claim 21 wherein themeans for receiving call admission policy information further comprisesa Call Admission Control Module (CAC) in communication with a switchingdevice in the network.